The present invention relates to the field of telecommunication networks, in particular, data latency introduced in fiber optics communication networks.
In a telecommunication network, messages generated by a source terminal pass through a network of links and nodes until they arrive at a destination terminal. A telecommunication network using fiber optics communications is able to provide voice, video and data services straight to the customers' homes or businesses.
Voice over Internet Protocol (Voice over IP, VoIP) is one of a family of internet technologies for delivery of voice communications and multimedia sessions over Internet Protocol (IP) networks, such as the internet. Real-time Transport Protocol (RTP) in conjunction with RTP Control Protocol (RTCP) is commonly used for delivering audio and video over IP networks. Voice telephony services, such as services normally associated with POTS (Plain Old Telephone Service), provided to the customers internally use a VoIP protocol for signaling such as Session Initiation Protocol (SIP).
RTP defines a standardized packet format for delivering audio and video over IP networks. Some non-limiting application examples of RTP include streaming media for communication and entertainment systems, such as, telephony and video conferencing applications. While RTP carries the media streams, for example, audio and video, RTCP is used to monitor transmission statistics and Quality of Service (QoS) and aids synchronization of multiple streams.
SIP is an IETF (Internet Engineering Task Force) defined signaling protocol, which is used for controlling multimedia communication sessions, such as voice and video calls, over IP networks. SIP, when used in conjunction with RTP, for voice and video stream communications, follows Session Description Protocol (SDP) for defining the parameters associated with the media streams. SDP does not deliver media itself but is used for negotiation between end points of media type, format, and all associated properties.
With an increasing number of users utilizing the networks for voice, data and video services, the network operators are facing the next big issue in ensuring the quality of service to the end users. In regards to the voice quality, the major problem that comes up is the voice transmission delay introduced by the network itself. This is further explained with the help of FIG. 1.
FIG. 1 illustrates a conventional telecommunication network 100.
As illustrated in FIG. 1, conventional telecommunication network 100 includes a user terminal 102 and a user terminal 104. Conventional telecommunication network 100 or just referred to as ‘network’ in this specification, is distributed into different network zones. In this example, a distributed network zone 106 represents the caller side and a distributed network zone 110 represents the callee side. A distributed network zone 108 includes a soft switch 118, which is the central intelligence for routing the calls.
User terminal 102 further includes a phone 126 and an optical network terminal (ONT) 128. User terminal 104 further includes a phone 132 and an ONT 130. Distributed network zone 106 further includes an optical line terminal (OLT) 112, a router 114, and a session border controller (SBC) 116. Distributed network zone 110 further includes a SBC 120, a router 122, and an OLT 124.
In FIG. 1, phone 126, ONT 128, phone 132, ONT 130, OLT 112, router 114, SBC 116, soft switch 118, SBC 120, router 122, and OLT 124 are illustrated as distinct devices. However, at least two of phone 126, ONT 128, phone 132, ONT 130, OLT 112, router 114, SBC 116, soft switch 118, SBC 120, router 122, and OLT 124 may be combined as a unitary device.
Conventional telecommunication network 100 makes use of RTP in conjunction with RTCP and SIP to provide voice, data and video services between user terminal 102 and user terminal 104. In this example, FIG. 1 illustrates a routing path for a voice packet traversing in conventional telecommunication network 100 from user terminal 102 to user terminal 104.
User terminal 102 may be a residence or an office building, where ONT 128 is installed. In FIG. 1, ONT 128 is connected to telephone 126, however, in other cases, ONT 128 may also be connected to a computer and/or a television set. ONT 128 operates as a media converter, which converts electric signals from phone 126 to fiber-optic light signals. ONT 128 is operable to deliver multiple POTs lines, internet data and video to OLT 112 via an optical link 134.
For the purposes of discussion, consider that a voice packet is to be transmitted from user terminal 102 to user terminal 104. The voice packet traverses through distributed network zone 106, distributed network zone 108 and distributed network zone 110 before it is received by user terminal 104.
ONT 128 transmits the voice packet in the form of an RTP packet via optical link 134, which is destined for SBC 116. The RTP packet employs the RTP protocol for real time data transfer and the RTCP protocol for control and QoS feedback and SIP signaling protocol. Each RTP packet includes an appropriate time stamp for synchronization, a sequence number for packet loss and reordering detection, and a payload format, which indicates the encoded format of the data, along with some other parameters, which are discussed in detail with the help of FIG. 2.
An RTP session is established, which includes a source and a destination IP address and a pair of ports for RTP and RTCP. The RTP packet is sent via optical link 134 and includes the destination IP address for SBC 116.
Optical link 134 may be a part of a Passive Optical Network (PON), which is terminated at OLT 112. OLT 112 can support a number of PONs, which are terminated at ONTs near end users. When OLT 112 receives the RTP packet via optical link 134 from ONT 128, the OLT coverts the RTP into electric signals.
OLT 112 transmits a signal 136 comprising the RTP packet to router 114, which functions as a gateway router for SBC 116. Router 114 works with SBC 116 in order to keep data flowing between networks or between different locations within distributed network zone 106. Router 114 forwards the RTP packet via a signal 138, which is destined for SBC 116.
SBC 116 functions as a gateway to interconnect distributed network zone 106 and distributed network zone 108 and manages flow of session data across distributed network zone 106 and distributed network zone 108. In addition, SBC 116 may perform security functions to protect the networks, quality of service (QoS) functions of a network, and connectivity functions among different parts of the network, and provides support for regulatory requirements. SBC 116 receives the RTP packet and maps it to an RTP packet destined for switch 118.
Switch 118 receives the RTP packet via a signal 140 from SBC 116. Switch 118 functions as a Back-to-Back User Agent (B2BUA), which is the central intelligence for routing calls. In one example, switch 118 may be a CS2K (Communication server 2000) or Broadsoft B2BUA softswitch. Switch 118 follows SIP to manage multimedia VoIP telephone calls. The switch operates between both end points of a phone call or communication session, which are phone 126 and phone 132 in this example, and divides the communication channels into two call legs and mediates all SIP signaling between both ends of the call, from call establishment to call termination. Switch 118 maps the RTP packet received from SBC 116 in to an RTP packet destined for SBC 120. Switch 118 delivers RTP packet destined for SBC 120 via a signal 142.
SBC 120 interfaces with distributed network zone 108 and sits at the border of distributed network zone 110. SBC 120 receives the RTP packet from switch 118 and maps the RTP packet to an RTP packet destined for ONT 130. SBC 120 transmits the RTP packet to router 122 via a signal 144.
Router 122 transmits RTP packet, which is destined for ONT 130, via a signal 146 to OLT 124.
OLT 124 coverts electric signals into fiber optic signals in order to deliver the RTP packet to ONT 130 via an optical link 148. ONT 130 coverts fiber optic signals to electric signals such that the voice packet transmitted originally by phone 126 can be received by phone 132.
In the illustrated example of FIG. 1, an active voice call between two POTS lines traverses via SBC 116, switch 118 and then back to SBC 120. SBC 116, switch 118 and SBC 120 act as a B2BUA and relay the RTP media. ONT 128, OLT 112, SBC 120, SBC 116, switch 118, OLT 124 and ONT 130 are also referred to as network elements. A conventional RTP packet generated by ONT 128 traverses a path as illustrated in FIG. 1, and is discussed below with the help of FIG. 2.
FIG. 2 illustrates a conventional RTP packet 200.
As illustrated in FIG. 2, conventional RTP packet 200 includes multiple data fields 202-216.
In this example of conventional RTP packet 200, data field 202 is shown to include a 2-bit version V, a padding bit P, an extension bit X, a 4-bit CSRC (Contributing Source) count CC, a marker bit M and a 7-bit payload type PT. However, in other cases, version bit V, padding bit P, extension bit X, CSRC count CC, marker bit M and payload type PT may correspond to individual fields.
Version V indicates the version of the RTP protocol. Padding P indicates if there are extra padding bytes at the end of the RTP packet. Padding may be used to fill up a block of certain size, for example, as required by encryption algorithms. Extension bit X indicates the presence of an extension header between the standard header and the payload data. CSRC count CC indicates the number of CSRC identifiers that follow the fixed header. CSRC identifiers enumerate contributing sources to a stream, which has been generated from multiple sources. Marker bit M is used at the application level and is defined by a profile. If M bit is set, it indicates that the current data has some special relevance for that application. Payload type PT indicates the format of the payload and determines its interpretation by the application.
Data field 204 indicates a sequence number, which is incremented by one for each RTP packet sent and is to be used by a receiver to detect packet loss and to restore packet sequence. Data field 206 indicates a timestamp, which is used by the receiver to play back the received samples at appropriate intervals. Data field 208 indicates a synchronization source (SSRC) identifier, which uniquely identifies a source of a stream of data. Data fields 202-208 are RTP fixed header fields. An extension mechanism is provided in order to allow additional information to be carried in the RTP data packet header using the extension bit X in the RTP header. If the extension bit X is set to one, the RTP fixed header must be followed by exactly one header extension, described below with the help of data fields 210-214.
When the extension bit X in the RTP header is set to one, data field 210, data field 212, and data field 214 are used to indicate any profile specific values for a specific application. Data field 210 indicates an extension header identifier (ID) specific to a profile. Data field 212 indicates extension header length, which indicates a length of the extension. Data field 214 indicates extension header itself.
Data field 216 comprises payload data transported by RTP in a packet, for example, audio samples or compressed video data.
As discussed with reference to FIG. 1, an RTP packet generated by ONT 128 traverses the path through distributed network zone 106, distributed network zone 108, and distributed network zone 110 before reaching ONT 130. Each network element will advertise its own IP address and port via Session Description Protocol (SDP) information exchanged as part of the Session Initiation Protocol (SIP) signaling when the RTP session is set up. When these network elements start receiving RTP and RTCP packets, they map these packets internally to another port and send it to the destination IP address and port. Hence, it is obvious that for data traversing from user terminal 102 to user terminal 104, there are three levels of RTP packet conversion being done, that is, at SBC 116, switch 118, and SBC 120. For a voice call made by user terminal 102, RTP packet conversion introduces latency in the voice as perceived by user terminal 104.
RTP packet conversion done at SBC 116, switch 118, and SBC 120 is performed to adhere to FBI wiretapping policies enforced in the networks. Ideally, a 50-100 millisecond (msec) end-to-end network delay introduced by the latency could be acceptable. Conventional telecommunication network 100 supports voice, data and video services for a huge number of ONTs deployed at the end users, which introduces considerable latency in conventional telecommunication network 100.
In situations when one or more SBCs or switches are taken off of the network, resulting in traffic that is overloaded on backup SBCs and switches, end-to-end network delay is higher due to longer latency. In such situations, when the end users suffer from delay in voice reception, there is currently no mechanism to troubleshoot the problem so as to find the highest latency path in the network and minimize it.
What is needed is a system and method to measure the latency introduced at each network element that participates as an RTP relay in a telecommunication network.